There’s been a lot of discussion lately around connecting Avaya (legacy Nortel) IP phones with third-party SIP capable call servers. I’ve personally toyed with Asterisk on a number of occasions and have always been impressed so I recently setup an Asterisk Now installation (AsteriskNOW-1.7.1-i386.iso) on a CentOS 6.2 KVM host so I could re-test the interoperability between the latest version of Asterisk (v1.6.2.20) and the 1100 and 1200 series IP phones from Avaya running SIP v4.0 and SIP v4.3 respectively.

The installation was pretty straight forward, however, there were a few small issues that I had to deal with. Initially I was unable to connect to the server and found that the firewall was enabled, so I had to disable the firewall with the following commands, service iptables stop, chkconfig iptables off. I was also getting a weird error in the FreePBX gui when I tried to apply the configuration;

exit: 126
sh: /var/lib/asterisk/bin/retrieve_conf: Permission denied

…this turned out to be an issue with SELINUX, so I had to edit /etc/selinux/config and disable SELINUX (a reboot is required for the change to take effect). Once I did those few steps I was ready to create some extensions so I created 1001 and 1002 and set their password (secret) to ‘abc123’.

The Avaya (legacy Nortel) IP phones can be provisioned from a TFTP server so I installed a TFTP server on my Asterisk server using yum install tftp-server. Then I enabled the TFTP server with chkconfig tftp on and finally I had to restart xinetd with service xinetd restart. I placed the files I needed in the /tftpboot directory including 1220SIP.cfg, 1120eSIP.cfg and users.dat (these filenames are case sensitive on a Linux server – if you use a Windows server such as TFTPD32 then the case is not an issue). I configured my local DHCP server to offer DHCP option 66 (TFTP Server) and I was off and running. The 1220 and 1120e both booted, download the provisioning files from the TFTP server, and connected to the Asterisk server. I entered the username and passwords and I was logged in and running in seconds placing calls between the two handsets.

I had to refer to my original post on the forums on what settings I needed to disable the extended license;

Over the past weekend I set out to setup Asterisk, an open source communication server, to test some of the issues reported in a thread over on the discussion forums. I had an Avaya 1120e and 1165e IP phone available to test with, however, both phones were running the UNIStim software for the Avaya Communications Server 1000. I needed to upgrade them to support SIP.

About a year ago I posted how I setup Asterisk to work with an i2002 IP phone utilizing the UNIStim channel driver. This time around I was looking to utilize the standard SIP channel driver with the 1120e and 1165e.

SIP Software

You’ll need to download the SIP software from the Avaya Support website. You should be able to retreive the SIP software from this link without needing to log into Avaya’s website. You should download the software for the appropriate model you’ll be working with. In my case I downloaded the following two files;

SIP1120e04.00.04.00.bin

SIP1165e04.00.04.00.bin

TFTP Server

You’ll need a TFTP server to host the files that the IP phone will download. You can use any TFTP server you already have on the network. If you don’t have a TFTP server you can use TFTPD32 from Philippe Jounin on any Microsoft Windows XP, Vista or Windows 7 personal computer. I download the zip and exploded the files to D:\Temp.

TFTP Files

With the TFTPD32 software in D:\Temp I then copied the two firmware images (SIP1120e04.00.04.00.bin and SIP1165e04.00.04.00.bin) to the same directory. At this point I needed to create some configuration (provisioning) files which the IP phones would download. The first file 1120e.cfg will be used for the 1120e IP phone;

You’ll need to substitute the IP address above (192.168.1.6) with the IP address of the personal computer that will be running TFTPD32. Now that you have all the files you’ll need for the upgrade, you can start the TFTPD32 executable. You should see a window similar to the figure to the right.

Upgrade

You need to make sure that the IP phones know which TFTP server to use. This can be accomplished via DHCP option 66 or it can be set in the device configuration on the actual IP phone itself. I was utilizing the DHCP server built into my Verizon FiOS router so I had to set the TFTP server manually via the IP phone configuration.

When you are ready just reboot the phone. As the IP phone boots up it will request an IP address from the DHCP server and it will check the TFTP serve. The IP phone should download the 1120e.cfg (or 1140e.cfg of 1165e.cfg depending on the model). Once the phone realizes there is a software update it will boot into BOOTPC mode in order to perform the actual upgrade.

You should see something similar to the following;

[FW] reading...
SIP1120e04.00.04.00.bin
VERSION SIP1120e04.00.04.00

Shortly followed by;

[FW] writing...
SIP1120e04.00.04.00.bin
VERSION SIP1120e04.00.04.00

Once the upgrade is complete the IP phone should reboot. I will warn you that you should I’ve seen some odd behavior between the settings on the IP phone and the settings that should be applied via the provisioning files. There have been a few cases where I needed to reconfigure the IP phone even though it appeared to be configured properly. In the few cases I’ve experienced reconfiguring the IP phone solved the problem.

Once the 1100 series IP phone is upgraded to SIP it will start looking for a new configuration file, 1120eSIP.cfg (or 1140eSIP.cfg or 1165eSIP.cfg depending on your model).

Avaya has released SIP software release 3.2 for their 1100 and 1200 series IP deskphones. This release adds support for the 1120e, 1140e, 1165e, 1220, and 1230 model IP deskphones.

Here are some of the enhancements made in the new software release;

Improved Licensing

SIP Support for 1220,1230 and 1165E IP Deskphones

Shared Call Appearances – CS1000

IPv6 Support

SRTP Media Security

TLS Signaling Security

Certificate-based Authentication

Enhanced Screensavers

Background images

Support for Avaya Aura™ Communication Manager / Session Manager

I was having a discussion with “Mike” in the comments section of any earlier post entitled, SIP Software Release 3.0 for IP Deskphones, in which he pointed out some of the issues with the new licensing model. Well it looks like Avaya was paying attention to that thread and made some changes to the licensing that should satisfy the majority of users. (I’m just going to quote directly from the readme.)

Improved Licensing

Licensing was introduced in the SIP 3.0 release. With SIP 3.2, the following changes are made to the licensing mechanism:

The Standard feature set is now available on all desksets without a token. This provides a basic set of SIP features conforming to RFC 3261 (SIPPING 19) at no additional cost.

Now, when the phone is registered to a recognized Avaya call server (Avaya AuraTM, AS 5300, CS1000 or CS2100), the Extended feature set is available as well without a token.

The Advanced feature set is reserved for Federal and DoD features on the AS 5300 call server only

The feature packages have been re-organized

Wideband is part of Standard feature set

IPv6 and Broadworks SCA are part of Extended feature set

Security is now part of the Extended feature set

If you connect your IP deskphone to a Avaya Call Server (Avaya AuraTM, AS 5300, CS1000 or CS2100), you’ll get all the standard features you would get with the UNIStim firmware. The licensing really only comes into play if you decide to connect your Avaya IP deskphone to a third party call server or SIP provider.

Avaya has released SIP software release 3.0 for their 1120E and 1140E IP deskphones. (There was no mention of the 1110E, 1150E, 1165E or 1200 series IP phones in any of the accompanying material).

Several enhancements have been included in SIP Release 3.0 for the 1100 series phones including User Interface and Preferences enhancements, Multi-user Login, Emergency Services support, USB device support, Wide-band Codec, Provisioning and Licensing.

The SIP software Release 3.0 for IP Deskphones also continues to improve the overall quality of the IP Deskphone software through the delivery of ongoing resolution of CRs. Numerous quality improvements have been delivered and 9 customer cases have been closed in SIP 3.0.

I’ve only performed very limited SIP testing with the 1120E, 1140E, and 1220 IP phones in non-production environments. I did notice a few feature called “Multi-user Login” which allows a SIP IP phone to connect to multiple SIP servers at the same time. Here’s the blurb from Avaya on the feature (it’s a direct quote from the release notes);

Multi-user Login

The Multiuser feature in SIP Release 3.0 allows multiple SIP user accounts to be in use on the IP Deskphone at the same time. Multiple users, each with their own account, can share a single IP Deskphone allowing each user to receive calls without logging off other users. One user can have multiple user accounts (for example, a work account and a personal account) active at the same time on the same IP Deskphone. You can register each account to a different server, and for each account, the IP Deskphone exposes the functionality available to that account. One account is considered a primary account and is used by default for most IP Deskphone operations. Each account is associated to a line key; the primary account is always on the bottom right line key of the IP Deskphone (this is the first key, Key 01), and an arbitrary key (including a key on an Expansion Module) can be selected for additional accounts.

The following operations are supported:

Start dialing

Place a call using the corresponding user account

Answer an incoming call targeted to that account

Initiate a call without pressing a line key (for example, by dialing digits at the idle screen and lifting the handset) uses the primary account.

A running IP Deskphone is associated to a single profile that represents one configuration of the IP Deskphone with all relevant persistent data such as preferences and call logs. A different profile is associated to each account used as a primary account. The IP Deskphone can store up to five different profiles; the IP Deskphone takes data from the profile associated to the current primary account. A number of configurations are independent of profiles and tied directly to an account making them available to that account regardless of the primary account you use (for example, voice mail ID). The IP Deskphone receives and answers calls targeted at any of the registered accounts; the incoming call screen indicates who the call is for. You can place an outgoing call using any of the accounts; the account that you use is displayed on the dialing screen. When a call is active, information from both local and remote parties appear on the screen. Regardless of which account receives the call, incoming call logs, outgoing call logs, and instant messages appear in a single list. The IP Deskphone indicates the local user in the detailed view of the entry.

Some features are only available to the primary account, such as instant messaging, retrieving parked calls by token, and establishing ad-hoc conference calls. Please refer to the product bulletin and the release notes for all the details.

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